EP3340660B1 - Binaurale filter für monophonkompatibilität und lautsprecherkompatibilität - Google Patents

Binaurale filter für monophonkompatibilität und lautsprecherkompatibilität Download PDF

Info

Publication number
EP3340660B1
EP3340660B1 EP18155721.6A EP18155721A EP3340660B1 EP 3340660 B1 EP3340660 B1 EP 3340660B1 EP 18155721 A EP18155721 A EP 18155721A EP 3340660 B1 EP3340660 B1 EP 3340660B1
Authority
EP
European Patent Office
Prior art keywords
filter
binaural
sum
signal
difference
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Active
Application number
EP18155721.6A
Other languages
English (en)
French (fr)
Other versions
EP3340660A1 (de
Inventor
Glenn N. Dickins
David S. Mcgrath
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Dolby Laboratories Licensing Corp
Original Assignee
Dolby Laboratories Licensing Corp
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Dolby Laboratories Licensing Corp filed Critical Dolby Laboratories Licensing Corp
Priority to EP23183853.3A priority Critical patent/EP4274263A3/de
Priority to EP20159771.3A priority patent/EP3739908B1/de
Publication of EP3340660A1 publication Critical patent/EP3340660A1/de
Application granted granted Critical
Publication of EP3340660B1 publication Critical patent/EP3340660B1/de
Active legal-status Critical Current
Anticipated expiration legal-status Critical

Links

Images

Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
    • H04S7/30—Control circuits for electronic adaptation of the sound field
    • H04S7/305—Electronic adaptation of stereophonic audio signals to reverberation of the listening space
    • H04S7/306—For headphones
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S5/00—Pseudo-stereo systems, e.g. in which additional channel signals are derived from monophonic signals by means of phase shifting, time delay or reverberation 
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S2400/00—Details of stereophonic systems covered by H04S but not provided for in its groups
    • H04S2400/01—Multi-channel, i.e. more than two input channels, sound reproduction with two speakers wherein the multi-channel information is substantially preserved
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S2400/00—Details of stereophonic systems covered by H04S but not provided for in its groups
    • H04S2400/03—Aspects of down-mixing multi-channel audio to configurations with lower numbers of playback channels, e.g. 7.1 -> 5.1
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S2420/00—Techniques used stereophonic systems covered by H04S but not provided for in its groups
    • H04S2420/01—Enhancing the perception of the sound image or of the spatial distribution using head related transfer functions [HRTF's] or equivalents thereof, e.g. interaural time difference [ITD] or interaural level difference [ILD]
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S3/00—Systems employing more than two channels, e.g. quadraphonic
    • H04S3/008—Systems employing more than two channels, e.g. quadraphonic in which the audio signals are in digital form, i.e. employing more than two discrete digital channels

Definitions

  • the present disclosure relates generally to signal processing of audio signals, and in particular to processing audio inputs for spatialization by binaural filters such that the output is playable on headphones, or monophonically, or through a set of speakers.
  • the audio input signals may be a single signal, a pair of signals for stereo reproduction, a plurality of surround sound signals, e.g., four audio input signals for 4.1 surround sound, five audio input signals for 5.1, seven audio input signals for 7.1, and so forth, and further might include individual signals for specific locations, like of a particular source of sound.
  • the binaural filters take into account the head related transfer functions (HRTFs) from each virtual speaker to each of a left ear and right ear, and further take into account both early echoes and the reverberant response of the listening room being simulated.
  • HRTFs head related transfer functions
  • US 2008/031462 describes techniques that can be used to provide methods of spatial audio rendering using adapted M-S matrix shuffler topologies.
  • Embodiments of the present invention includes a method, an apparatus, and program logic, e.g., program logic encoded in a computer readable medium that when executed cause carrying out of the method.
  • program logic e.g., program logic encoded in a computer readable medium that when executed cause carrying out of the method.
  • the dependent claims relate to preferred embodiments.
  • One example of a method is of processing one or more audio input signals for rendering over headphones using binaural filters to achieve virtual spatializing of the one or more audio inputs with the additional the property that the binauralized signals sound good when played back monophonically after downmixing or when played back through relatively closely spaced loudspeakers.
  • Another example of a method is of operating a data processing system for processing one or more pairs of binaural filter characteristics, e.g., binaural filter impulse responses to determine corresponding one or more pairs of modified binaural filter characteristics, e.g., modified binaural filter impulse responses, so that when one or more audio input signals are binauralized by respective one or more pairs of binaural filters having the one or more pairs of modified binaural filter characteristics, the binauralized signals achieve virtual spatializing of the one or more audio inputs with the additional property that the binauralized signals sound good when played back monophonically after downmixing or over relatively closely spaced loudspeakers.
  • binaural filter impulse responses e.g., aural filter impulse responses
  • modified binaural filter impulse responses e.g., modified binaural filter impulse responses
  • the apparatus includes a pair of binaural filters characterized by one or more pairs of base binaural filters, with one pair of base binaural filters for each of the audio signal inputs.
  • Each pair of base binaural filters is representable by a base left ear filter and a base right ear filter, and further representable by a base sum filter and a base difference filter.
  • Each filter is characterizable by a respective impulse response.
  • At least one pair of base binaural filters is configured to spatialize its respective audio signal input to incorporate a direct response to a listener from a respective virtual speaker location, and to incorporate both early echoes and a reverberant response of a listening room.
  • the apparatus generated output signals that are playable either through headphones or monophonically after a monophonic mix.
  • the transition of the base sum filter impulse response to an insignificant level occurs gradually over time in a frequency dependent manner over an initial time interval of the base sum filter impulse response.
  • the base sum filter decreases in frequency content from being initially full bandwidth towards a low frequency cutoff over the transition time interval.
  • the transition time interval is such that the base sum filter impulse response transitions from full bandwidth up to about 3ms to below 100Hz at about 40ms.
  • the base difference filter length at high frequencies of above 10 kHz is less than 40ms, the base difference filter length at frequencies of between 3 kHz and 4 kHz, is less 100ms, and at frequencies less than 2 kHz, the base difference filter length is less than 160ms.
  • the base difference filter length at high frequencies of above 10 kHz is less than 20ms, the base difference filter length at frequencies of between 3 kHz and 4 kHz, is less 60ms, and at frequencies less than 2 kHz, the base difference filter length is less than 120ms.
  • the base difference filter length is less than about 800ms. In some of these examples, the base difference filter length is less than about 400ms. In some of these examples, the base difference filter length is less than about 200ms.
  • the base sum filter length decreasing with increasing frequency
  • the base sum filter length for all frequencies less than 100 Hz is at least 40 ms and at most 160 ms
  • the base sum filter length for all frequencies between 100 Hz and 1 kHz is at least 20 ms and at most 80 ms
  • the base sum filter length for all frequencies between 1 kHz and 2 kHz is at least 10 ms and at most 20 ms
  • the base sum filter length for all frequencies between 2 kHz and 20 kHz is at least 5ms and at most 20 ms.
  • the base sum filter length for all frequencies less that 100 Hz is at least 60 ms and at most 120 ms
  • the base sum filter length for all frequencies between 100 Hz and 1 kHz is at least 30 ms and at most 60 ms
  • the base sum filter length for all frequencies between 1 kHz and 2 kHz is at least 15 ms and at most 30 ms
  • the base sum filter length for all frequencies between 2 kHz and 20 kHz is at least 7ms and at most 15 ms.
  • the base sum filter length for all frequencies less that 100 Hz is at least 70 ms and at most 90 ms
  • the base sum filter length for all frequencies between 100 Hz and 1 kHz is at least 35 ms and at most 50 ms
  • the base sum filter length for all frequencies between 1 kHz and 2 kHz is at least 18 ms and at most 25 ms
  • the base sum filter length for all frequencies between 2 kHz and 20 kHz is at least 8ms and at most 12 ms.
  • the base binaural filter characteristics are determined from a pair of to-be-matched binaural filter characteristics.
  • the base difference filter impulse response is at later times substantially proportional to the difference filter of the to-be-matched binaural filter.
  • the base difference filter impulse response becomes after 40 ms substantially proportional to the difference filter of the to-be-matched binaural filter.
  • Particular examples include a method of binauralizing a set of one or more audio input signals.
  • the method comprises filtering the set of audio input signals by a binauralizer characterized by one or more pairs of base binaural filters.
  • the base binaural filters in different examples, are as described in above in this Overview Section in describing particular apparatus examples.
  • Particular examples include a method of operating a signal processing apparatus.
  • the method includes accepting a pair of signals representing the impulse responses of a corresponding pair of to-be-matched binaural filters configured to binauralize an audio signal, and processing the pair of accepted signals by a pair of filters each characterized by a modifying filter that has time varying filter characteristics.
  • the processing forms a pair of modified signals representing the impulse responses of a corresponding pair of modified binaural filters.
  • the modified binaural filters are configured to binauralize an audio signal and further have the property that of a low perceived reverberation in a monophonic mix down, and minimal impact on the binaural filters over headphones.
  • the modified binaural filters are characterizable by a modified sum filter and a modified difference filters.
  • the time varying filters are configured such that modified binaural filters impulse responses include a direct part defined by head related transfer functions for a listener listening to a virtual speaker at a predefined location.
  • the modified sum filter has a significantly reduced level and a significantly shorter reverberation time compared to the modified difference filter, and there is a smooth transition from the direct part of the impulse response of the sum filter to the negligible response part of the sum filter, with smooth transition being frequency selective over time.
  • Particular examples include a method of operating a signal processing apparatus.
  • the method includes accepting a left ear signal and right ear signal representing the impulse responses of corresponding left ear and right ear binaural filters configured to binauralize an audio signal.
  • the method further includes shuffling the left ear signal and right ear signal to form a sum signal proportional to the sum of the left and right ear signals and a difference signal proportional to difference between the left ear signal and the right ear signal.
  • the method further includes filtering the sum signal by a sum filter that has time varying filter characteristics, the filtering forming a filtered sum signal, and processing the difference signal by a difference filter that is characterized by the sum filter, the processing forming a filtered difference signal.
  • the method further includes unshuffling the filtered sum signal and the filtered difference signal to form modified a modified left ear signal and modified right ear signal representing the impulse responses of corresponding left ear and right ear modified binaural filters.
  • the modified binaural filters are configured to binauralize an audio signal, are representable by a modified sum filter and a modified difference filters.
  • the modified binaural filters have the properties of the base binaural filters described above in this Overview Section for the particular apparatus examples.
  • Particular examples include program logic that when executed by at least one processor of a processing system causes carrying out any of the method examples described above in this Overview Section for the particular apparatus examples.
  • Particular examples include a computer readable medium having therein program logic that when executed by at least one processor of a processing system causes carrying out any of the method examples described above in this Overview Section for the particular apparatus examples.
  • the apparatus comprises a processing system that has at least one processor, and a storage device.
  • the storage device is configured with program logic that causes when executed the apparatus to carry out any of the method examples described above in this Overview Section for the particular apparatus examples.
  • FIG. 1 shows a simplified block diagram of a binauralizer 101 that includes a pair of binaural filters 103, 104 for processing a single input signal. While binaural filters are generally known in the art, binaural filters that include the monophonic playback features described herein are not prior art.
  • u ( t ) a single audio signal to be binauralized by the binauralizer 101 for binaural rendering through headphones 105
  • h L ( t ) and h R ( t ) respectively, the binaural filter impulse responses for the left and right ear, respectively, for a listener 107 in a listening room.
  • the binauralizer is designed to provide to the listener 105 the sensation of listening to the sound of signal u(t) coming from a source-a "virtual loudspeaker" 109 at a pre-defined location.
  • signals that have been binauralized for headphone use may be available.
  • the binauralization processing of the signals may be by one or more pre-defined binaural filters that are provided so that a listener has the sensation of listening to content in different type of rooms.
  • One commercial binauralization is known as DOLBY HEADPHONE (TM).
  • TM DOLBY HEADPHONE
  • the binaural filters pairs in DOLBY HEADPHONE binauralization have respective impulse responses with a common non-spatial reverberant tail.
  • DOLBY HEADPHONE implementations offer only a single set of binaural filters describing a single typical listening room, while other can binauralize using one of three different sets of binaural filters, denoted DH1, DH2, and DH3. These have the following properties:
  • ⁇ the convolution operation by ⁇
  • ⁇ the convolution of a(t) and b(t)
  • time dependence is not explicitly shown on the left hand side, but would be implied by the use of a letter. Non-time dependent quantities will be clearly indicated.
  • a binaural output includes a left output signal denoted v L (t) and a right ear signal denoted v R (t).
  • FIG. 1 shows a single input audio signal.
  • FIG. 2 shows a simplified block diagram of a binauralizer that has one or more audio input signals denoted u 1 ( t ) , u 2 ( t ) , ... u M ( t ) , where M is the number of input audio signals.
  • M can be one, or more than 1.
  • the binaural filters take into account the respective head related transfer functions (HRTFs) for each virtual speaker location and left and right ears, and further take into account both early echoes and reverberant response of the listening room being simulated.
  • HRTFs head related transfer functions
  • the left and right binaural filters for the binauralizer shown include left ear binauralizers and right each binauralizers 203-1 and 204-1, 203-2 and 204-2, ...., 203- M and 204- M having impulse responses h 1 L ( t ) and h 1 R ( t ) , h 2L ( t ) and h 2 R ( t ) , ..., h ML ( t ) and h MR ( t ) , respectively.
  • the left ear and right ear outputs are added by adders 205 and 206 to produce outputs v L (t) and v R (t).
  • the number of virtual speakers is denoted by M v .
  • upmixing may be incorporated to spatialize a pair of stereo input signals to sound to the listener on headphones as if there are five virtual loudspeakers.
  • FIG. 3 shows a simplified block diagram of a binauralizer 303 having one or more audio input signals and generating a left output signal v L (t) and a right ear signal denoted v R (t).
  • v M (t) a monophonic mix down of the left and right output signals obtained by down-mixer 305 that carries out some filtering on each of the left and right signals v L (t) and a right ear signal denoted v R (t) and adds, i.e., mixes the filtered signals.
  • v M (t) a monophonic mix down of the left and right output signals obtained by down-mixer 305 that carries out some filtering on each of the left and right signals v L (t) and a right ear signal denoted v R (t) and adds, i.e., mixes the filtered signals.
  • the description that follows assumes a single input u(t).
  • m L ( t ) and m R ( t ) the impulse responses of the filters 307 and 308 on the left and right output signals, respectively, of the down-mixer 305.
  • u u
  • ⁇ some scale factor constant.
  • the desired result is that m L ⁇ h L + m R ⁇ h R -each impulse response being a discrete function-is proportional to a unit impulse response.
  • h L ( t ) and h R ( t ) provide good binauralization, i.e., that the rendering of the outputs sounds natural via headphones as if the sound is from the virtual speaker location(s) and in a real listening room. It is further desirable that the monophonic mix of the binaural outputs when rendered sounds like the audio input u(t).
  • FIG. 4A shows a simplified block diagram of a shuffling operation by a shuffler 401 on a left ear stereo signal u L ( t ) and a right ear stereo signal u R ( t ) , followed by a sum filter 403 and a difference filter 404 having sum filter impulse response and difference filter impulse response h S ( t ) and h D ( t ) , respectively, followed by a de-shuffler 405, essentially a shuffler and a halver of each signal, to produce a left ear binaural signal output v L ( t ) and a right ear binaural signal output v R ( t ).
  • FIG. 4B shows simplified block diagram of a shuffling operation by the shuffler 401 on a left ear binaural filter impulse response h L ( t ) and a right ear binaural filter impulse response h R ( t ) to generate the sum filter binaural impulse response h S ( t ) and the difference filter binaural impulse response h D ( t ) .
  • de-shuffling by the de-shuffler 405, essentially a shuffler and a halver, to give back the left ear binaural filter impulse response h L ( t ) and the right ear binaural filter impulse response h R ( t ) .
  • Particular embodiments of the invention include a method of operating a signal processing apparatus to modify a provided pair of binaural filter characteristics to determine a pair of modified binaural filter characteristics.
  • One embodiment of the method includes accepting a pair of signals representing the impulse responses of a corresponding pair of binaural filters that are configured to binauralize an audio signal.
  • the method further includes processing the pair of accepted signals by a pair of filters each characterized by a modifying filter that has time varying filter characteristics, the processing forming a pair of modified signals representing the impulse responses of a corresponding pair of modified binaural filters.
  • the modified binaural filters are configured to binauralize an audio signal to a pair of binauralized signals and further have the property that a monophonic mix of the binauralized signals sounds natural to a listener.
  • the monophonic filters are simple to implement, and preferably compatible with general practice for monophonic down mixing of stereo content.
  • h S ( t ) 0 for t >0 .
  • FIG. 5 shows in simplified form a typical binaural filter impulse response, say for the sum filter h S ( t ) or for either the left or right ear binaural filter.
  • the general form of such an acoustical impulse response includes the direct sound, some early reflections, and a later part of the response consisting of closely spaced reflections and thus well approximated by a diffuse reverberation.
  • One aspect of the invention is a set of binaural filters defined by impulse responses h L ( t ) and h R ( t ) that also provide satisfactory binauralization, e.g., similar to a set of given filters h L 0 ( t ) and h R 0 ( t ) , but whose outputs also sound good when mixed down to a monophonic signal.
  • h L ( t ) and h R ( t ) compare to h L 0 ( t ) and h R 0 ( t ) , and how would one design h L ( t ) and h R ( t ) given h L 0 ( t ) and h R 0 ( t ).
  • the direct response encodes the level and time differences to the two respective ears which is primarily responsible for the sense of direction imparted to the listener.
  • HRTF direct head related transfer function
  • a typical HRTF also includes a time delay component. That means that when the binauralized outputs are mixed to a monophonic signal, the equivalent filter for the monophonic signal will not be minimum phase and will introduce some additional spectral shaping.
  • these delays are relatively short, e.g., ⁇ 1 ms.
  • the direct portions of the binaural filter impulse response of h L ( t ) and h R ( t )-those defined by the HRTFs- are the same as for any binaural filter impulse response, e.g., of filters h L 0 ( t ) and h R 0 ( t ) . That is, the characteristics of the binaural filters h L ( t ) and h R ( t ) that are looked at according to some aspects of the invention exclude the direct part of the impulse responses of the binaural filters.
  • this spectral shaping is taken into account.
  • one embodiment includes a compensating equalization filter to achieve a flatter spectral response. This is often referred to as compensating for the diffuse field head response, and how to carry such filtering would be straightforward to those in the art. Whilst such compensation can remove some of the spectral binaural cues, it does lead to spectral colouration.
  • the difference channel In order to maintain approximately the same energy in the sum and difference filters, the difference channel should be boosted by about 3dB compared to the original filter if required to maintain the correct spectrum and ratio of direct to reverberant energy in the modified responses.
  • this modification causes an undesirable degradation of the binaural imaging.
  • the sudden change in the interaural cross correlation has a strong perceptual effect, and destroys much of the sense of space and distance.
  • h D t h D 0 t for small values of t , say t ⁇ 3 ms
  • h D t 2 h D 0 t for large values of t , 2 . g ⁇ . , t > 40 ms .
  • the binaural filters have a difference filter impulse response that is a 3dB boost of a typical binaural difference filter impulse response for the direct part of the impulse response, e.g., ⁇ 3 ms, and have a flat constant value impulse response in the later part of the reverberant part of the difference filter impulse response.
  • the sudden change in the interaural cross correlation has a strong perceptual effect, and destroys much of the sense of space and distance.
  • One aspect of this disclosure is the introducing monophonic compatibility constraint in the later part of the binaural response in a gradual way that is perceptually masked, and thus has minimal impact on the binaural imaging.
  • the sum filter of the binaural pair is related to a typical sum filter of a typical binaural filter pair by a time-varying filter.
  • f ( t, ⁇ ) the time varying impulse response of the time varying filter
  • f ( t, ⁇ ) is or approximates a zero delay, linear phase, low pass filter impulse response with decreasing time dependent bandwidth denotes by ⁇ ( t ) >0, such that the time dependent frequency response, denoted
  • the time varying bandwidth is monotonically decreasing in time, i.e., ⁇ t 1 > ⁇ t 2 for t 1 ⁇ t 2 .
  • the difference filter impulse response is, at later times, e.g., after 40 ms, proportional to the difference filter of the to-be-matched or typical binaural filter.
  • the target binaural filters can then be reconstructed using the shuffling relationship of Eqs. (8a) and (9a) and FIG. 4B , or of Eqs. (8b) and (9b).
  • This approach has been found to provide an effective balance between reverberation reduction in the monophonic mix down, and perceptually masked impact on the binaural response.
  • the transition to a correlation coefficient of -1 occurs smoothly, and during an initial time interval, e.g., initial 40 ms of the impulse responses.
  • the reverberant response in the monophonic mix down is restricted to around 40 ms, with the high frequency reverberation being much shorter.
  • the 40 ms time is suggested for the monophonic mix down to be almost perceptually anechoic. Although some early reflections and reverberation may still exist in the monophonic mix, this is effectively masked by the direct sound and the inventor has found is not perceived as a discrete echo or additional reverberation.
  • the invention is not limited to the length 40 ms of the transition region. Such transition region may be altered depending on the application. If it is desired to simulate a room with a particularly long reverberation time, or low direct to reverberation ratio, the transition time could be extended further and still provide an improvement to the monophonic compatibility compared to standard binaural filters for such a room.
  • the 40 ms transition time was found to be suitable for a specific application where the original binaural filters had a reverberation time of 150 ms and the monophonic mix was required to be as close to anechoic as possible.
  • the sum filter is completely eliminated, this is not a requirement.
  • the magnitude of the sum impulse response is reduced by a factor sufficient to achieve a noticeable difference or reduction in the reverberation part of the monophonic mix down.
  • the inventor chose as a criterion the "just noticeable difference" for changes in reverberation level of around 6 dB.
  • a reduction in the sum filter reverberation response of at least 6dB is used compared to what occurs with a monophonic mix down of signals binauralized with typical binaural filters.
  • the sum filter is not completely eliminated, but its influence, e.g., the magnitude of its impulse response is significantly reduced, e.g., by attenuating the sum channel filter impulse response amplitude by 6dB or more.
  • a typical value for ⁇ is 1/2, which weights the original and modified sum filter impulse responses equally. In alternate embodiments, other weighting are used.
  • FIG. 6 shows a simplified block diagram of signal processing apparatus
  • FIG. 7 shows a simplified flowchart of a method of operating a signal processing apparatus.
  • the apparatus is to determine a set of a left ear signal h L ( t ) and a right ear signal h L ( t ) that form the left ear and right ear impulse responses of a binaural filter pair that approximates the binauralizing of a binaural filter pair that has left ear and right rear impulse responses h L 0 ( t ) and h R 0 ( t ) .
  • the method includes in 703 accepting a left ear signal h L 0 ( t ) and right ear signal h R 0 ( t ) representing the impulse responses of corresponding left ear and right ear binaural filters configured to binauralize an audio signal and whose binaural response is to be matched.
  • the method further includes in 705 shuffling the left ear signal and right ear signal to form a sum signal proportional to the sum of the left and right ear signals and a difference signal proportional to difference between the left ear signal and the right ear signal. In the apparatus of FIG. 6 , this is carried out by shuffler 603.
  • the method further includes in 707 filtering the sum signal by a time varying filter (a sum filter) 605 that has time varying filter characteristics, the filtering forming a filtered sum signal, and processing the difference signal by a different time varying filter 607-a difference filter-that is characterized by the sum filter 605, the processing forming a filtered difference signal.
  • the method further includes in 709 un-shuffling the filtered sum signal and the filtered difference signal to form to produce a left ear signal and a right ear signal proportional respectively to left and right ear impulse responses of binaural filters whose spatializing characteristics match that of the to-be-matched binaural filters, and whose outputs can be down-mixed to a monophonic mix with acceptable sound.
  • a time varying filter a sum filter
  • the de-shuffler 609 is the same as the shuffler 603 with an added divide by 2.
  • the resulting impulse responses define binaural filters configured to binauralize an audio signal and further have the property that the sum channel impulse response decreases smoothly to an imperceptible level, e.g., more than -6dB in the first 40 ms or so and the difference channel transitions to become proportional to a typical or particular to-be-matched binaural filter difference channel impulse response in the in the first 40 ms or so.
  • the method includes accepting a pair of signals representing the impulse responses of a corresponding pair of binaural filters configured to binauralize an audio signal.
  • the method includes processing the pair of accepted signals by a pair of filters each characterized by a modifying filter that has time varying filter characteristics, the processing forming a pair of modified signals representing the impulse responses of a corresponding pair of modified binaural filters.
  • the modified binaural filters are configured to binauralize an audio signal and further have the property that of a low perceived reverberation in the monophonic mix down, and minimal impact on the binaural filters over headphones.
  • the binaural filters according to one or more aspects of the present invention have the properties of:
  • the output signals binauralizer with filters according to an embodiment of the invention are also compatible with playback over a set of loudspeakers.
  • Acoustical cross-talk is the term used to describe the phenomenon that when listening to a stereo pair of loudspeakers, e.g., at approximately center front of a listener, each ear of the listener will receive signal from both of the stereo loudspeakers.
  • the acoustical cross talk causes some cancellation of the lower frequency reverberation.
  • the later parts of a reverberant response to an input become progressively low pass filtered.
  • signals binauralized with filters binaural filters according to embodiments of the present invention have been found to sound less reverberant when auditioned over speakers. This is particularly the case small relatively closely spaced stereo speakers, such as may be found in a mobile media device.
  • binaural filters that involve relatively less computation to implement by using the observation that the reverberation part of an impulse response is less sensitive to spatial location.
  • many binaural processing systems use binaural filters whose impulse responses have a common tail portion for the different simulated virtual speaker positions. See for example, above-mentioned patent publications WO 9914983 and WO 9949574 .
  • Embodiments of the present invention are applicable to such binaural processing systems, and to modifying such binaural filters to have monophonic playback compatibility.
  • binaural filters designed according to some embodiments of the present invention have the property that the late part of the reverberant tails of the left and right ear impulse responses are out of phase, mathematically expressed as h R ( t ) ⁇ - h L ( t ) for time t> 40 ms or so. Therefore, according to a relatively low computational complexity implementation of the binaural filters, only a single filter impulse response need be determined for the later part of the response, and such determined late part impulse response is usable in each of the left and right ear impulse responses of binaural filter pairs for all virtual speaker locations, leading to savings in memory and computation.
  • the sum filter of each such binaural filter pair includes a gradual time varying frequency cut off which extends the sum filter low frequency content further into the binaural response.
  • FIG. 8 shows a portion of code in the syntax of MATLAB (Mathworks, Inc., Natick, Massachusetts) that carries out part of the method of converting a pair of binaural filter impulse responses to signals representative of impulse responses of binaural filters.
  • the linear phase, zero delay, time varying low pass filter is implemented using a series of concatenated first order filters. This simple approach approximates a Gaussian filter.
  • This brief section of MATLAB code takes a pair of binaural filters h_L0 and h_R0, and creates a set of output binaural filters h_L and h_R. It is based on a sampling rate of 48kHz.
  • the input filters are shuffled to create the original sum and difference filter. (see lines 1-2 of the code)
  • the 3dB bandwidth of the Gaussian filter (B) is varied with the inverse square of the sample number and appropriate scaling coefficients. From this the associated variance of the Gaussian filter is calculated (GaussVar), and divided by four to obtain the variance of the exponential first order filter (ExponVar). In 805, this is used to calculate the time varying exponential weighting factor (a). (See lines 3-6 of the code).
  • the filter is implemented in 807 using two forward and two reverse passes of the first order filter. Both the sum and difference responses are filtered. (See lines 7-12 of the code).
  • the difference recreated from a scaled up version of the original difference response, less an appropriate amount of the filtered difference response. This is in effect a frequency selective boost of the difference channel from 0dB at time zero to +3dB in the later response. (See line 13 of the code).
  • FIG. 9 shows a plot of the impulse response of the time varying filter f ( t, ⁇ ) to an impulses at several times ⁇ : at 1, 5, 10, 20 and 40 ms. The first two impulses are beyond the vertical scale of the figure.
  • FIG. 9 clearly shows the Gaussian approximation of the applied filter impulse response and the increasing variance of the approximately Gaussian filter impulse response with time. Since the first order filter is run both forward and backwards, the resulting filter approximates a zero delay, linear phase, low pass filter.
  • FIG. 10 shows plots of the frequency response energy of the time varying filter of impulse response f ( t, ⁇ ) at times ⁇ of 1, 5, 10, 20 and 40 ms. It can be seen that the direct part of the response, in this case approximately from 0 to 3 ms, will be largely unaffected by the filter, whilst by 40 ms the filter causes almost 10dB of attenuation down to 100Hz. Because of the approximately Gaussian shape of the impulse response, the frequency response also has an approximately Gaussian profile. This approximately Gaussian frequency response profile, and the variation of the cut off frequency over time both help to achieve the perceptual masking of the modification made to the original filter.
  • FIG. 11 shows the original left ear impulse response h L0 ( t ) and modified left ear impulse response h L ( t ) . It is evident that both have a similar level of reverberant energy.
  • the direct sound remains unchanged. Note that the initial impulse of the direct sound measures around 0.2 and cannot be shown on the scale in the figure.
  • FIG. 12 shows a comparison of the original and modified summation impulse responses response h S0 ( t ) and h S ( t ) This clearly demonstrates the reduced level and reverberation time of the summation response. This is the characteristic that achieves a significant reduction in the reverberation when the output is mixed down to monophonic. It can also be seen that the modified summation response h S ( t ) becomes progressively low pass filtered, with only the lowest frequency signal components extending beyond the early part of the response.
  • FIG. 13 shows the original and modified difference impulse responses h D0 ( t ) and h D ( t ) . It can be observed that the difference signal is boosted in level. This is to achieve comparable spectra of the two responses.
  • the binaural filters when used to filter a source signal, e.g., by convolving with the binaural impulse response or otherwise applied to a source signal, add a spatial quality that simulates direction, distance and room acoustics to a listener listening via headphones.
  • Time-frequency analysis e.g., using the short time Fourier transform or other short time transform on sections signals that may overlap is well known in the art.
  • frequency-time analysis plots are known as spectrograms.
  • a short time Fourier transform e.g., in typically implemented as a windowed discrete Fourier transform (DFT) over a segment of a desired signal.
  • DFT discrete Fourier transform
  • Other transforms also may be used for time-frequency analysis, e.g., wavelet transforms and other transforms.
  • An impulse response is a time signal, and hence may be characterized by its time-frequency properties.
  • the inventive binaural filters may be described by such time-frequency characteristics.
  • the binaural filters according to one or more aspects of the present invention are configured to achieve simultaneously a convincing binaural effect over headphones, e.g., according to a pair of to-be-matched binaural filters, and a monophonic playback compatible signal when mixed down to a single output.
  • Binaural filter embodiments of the invention are configured to have the property that the (short time) frequency response of the binaural filter impulse responses varies over time with one or more features.
  • the sum filter impulse response e.g., the arithmetic sum of the two left and right binaural filter impulse responses, has a pattern over time and frequency that differs significantly from the difference filter impulse response, e.g., the arithmetic difference of the left and right binaural filter impulse responses.
  • the sum and difference filters show a very similar variation in frequency response over time.
  • the early part of the response contains the majority of the energy, and the later response contains the reverberant or diffuse component. It is the balance between the early and late parts, and the characteristic structure of the filters that imparts the spatial or binaural characteristics of the impulse response.
  • this reverberant response usually degrades the signal intelligibility and perceived quality.
  • FIGS. 14A-14E show plots of the energy as a function of frequency in the sum and difference filter responses at varying time spans along the length of the filter. While arbitrary, the inventor selected the time slices of 0-5 ms, 10-15 ms, 20-25 ms, 40-45 ms and 80-85 ms for this description. The 5 ms span of each section is to maintain a consistent length for comparative power levels, and it is also sufficient to capture some of the echoes and details in the filters, which can be sparse over time.
  • FIGS. 14A-14E show the frequency spectra for 5 ms segments at these times for a typical pair, for a simplistic monophonic compatibility pair, and for new binaural filter pair according to one or more aspects of the invention.
  • the impulse responses of simplistic monophonic compatibility pair were determined from the typical (to-be-matched pair). Furthermore, the impulse responses of the filters that include features of the present invention were determined from the typical (to-be-matched pair) according to the method described hereinabove.
  • the frequency energy response was calculated using the short time Fourier transform as a short-time windows DFT. No overlap was used for determine the five sets of frequency responses.
  • FIG. 14A for the first 5 ms starting at time 0 ms, it can be seen that the three responses are almost identical. This is the very early part of the response that is based on the HRTF from a virtual speaker location to impart a sense of direction. Any spread of the signal or echoes in the filter in this time are largely perceptually ignored due to the masking effect and dominant initial impulse.
  • the sum filter of the novel filter pair is further attenuated with the bandwidth coming down to around 1kHz.
  • the difference filter of the novel filter pair is boosted to maintain a similar binaural level and frequency response overall to that of a typical or to-be-matched filter pair.
  • a set of binaural filters is proposed with a shaping of the binaural filter impulse responses configured to achieve very good monophonic playback compatibility.
  • the filters are configured such that the monophonic response is constrained to the first 40 ms.
  • filter extent and “filter length” is the point at which the impulse response of the filter falls below -60dB of its initial value. This is also known in the art as the "reverberation time.”
  • the overall extent, e.g., the reverberation of the difference filter should not be too long.
  • the inventor has found that a reverberation time of 200ms produces excellent results, 400ms produces acceptable results, while the audio starts to sound problematic with a filter length of 800ms.
  • Table 1 provides a set of typical values for the sum filter impulse response lengths for different frequency bands, and also a range of values of the sum filter impulse response length for the frequency bands which still would provide a balance between monophonic playback compatibility and listening room spatialization.
  • Table 1 Frequency band (bandwidth) Typical sum filter length Range of sum filter lengths 0-100 Hz 80 ms 40-160 ms 100-1 kHz 40 ms 20-80 ms 1-2 kHz 20 ms 10-40 ms 2-20 kHz 10 ms 5-20 ms
  • time dependent frequency shaping depends on the nature and reverberance of the desired binaural response, e.g., as characterized by a set of to-be-matched binaural filters h L 0 ( t ) and h R 0 (t) as described hereinabove, and also on the preference for clarity in the monophonic mix against the approximation or constraint in the binaural filters.
  • FIGS. 15A and 15B show equal attenuation contours on the time-frequency plane for the sum and frequency filter impulse responses, respectively of an example binaural filter pair embodiment
  • FIGS. 16A and 16B show isometric views of the surface of the time-frequency plots, i.e., of spectrograms.
  • the contour data was obtained by using the windowed short time Fourier transform on 5 ms long segments that start 1.5 ms apart, i.e., that have significant overlap.
  • FIGS. 17A and 17B show the same isometric views of the surface of the time-frequency plots as FIGS. 16A and 16B , but for the sum and frequency filter impulse responses, respectively of a typical binaural filter pair, in particular, the binaural filters that those used for FIGS. 16A and 16B are to match. Note that in a typical binaural filter pair, the shape of the time-frequency plots of the sum and difference filters' respective impulse responses are not that different.
  • FIGS. 15A, 15B, 16A, 16B, 17A, and 17B in order to simplify the drawings so as not to obscure features of the time-frequency characteristics with small-detail variations in the respective responses.
  • the to-be-matched impulse response has a binaural response with a 200-300 ms reverberation time, and corresponds to DOLBY HEADPHONE DH3 binaural filters. There were no statistical significant cases in which the subjects preferred one binaural response over the other in the test. However the monophonic mix was substantially improved and unanimously preferred by all subjects for all source material tested.
  • binaural filters are not only applicable for binaural headphone playback, but may be applied to stereo speaker playback.
  • crosstalk between the left and right ear of a listener during listening, e.g., crosstalk between the output of a speaker and the ear furthest from the speaker.
  • crosstalk refers to the left ear hearing sound from the right speaker, and also to the right ear hearing sound from the left speaker.
  • the crosstalk essentially causes the listener to hear the sum of the two speaker outputs. This is essentially the same as monophonic playback.
  • the digital filters may be implemented by many methods.
  • the digital filters may be carried out by finite impulse response (FIR) implementations, implementations in the frequency domain, overlap transform methods, and so forth. Many such methods are known, and how to apply them to the implementations described herein would be straightforward to those in the art.
  • FIR finite impulse response
  • FIG. 18 shows a form of implementation of an audio processing apparatus for processing a set of audio input signals according to aspects of the invention.
  • the audio processing system includes: an input interface block 1821 that include an analog-to-digital (A/D) converter configured to convert analog input signals to corresponding digital signals, and an output block 1823 with a digital to analog (D/A) converter to convert the processed signals to analog output signals.
  • the input block 1821 also or instead of the A/D converter includes a SPDIF (Sony/Philips Digital Interconnect Format) interface configured to accept digital input signals in addition to or rather than analog input signals.
  • the apparatus includes a digital signal processor (DSP) device 1800 capable of processing the input to generate the output sufficiently fast.
  • DSP digital signal processor
  • the DSP device includes interface circuitry in the form of serial ports 1817 configured to communicate the A/D and D/A converters information without processor overhead, and, in one embodiment, an off-device memory 1803 and a DMA engine 1813 that can copy data from the off-chip memory 1803 to an on-chip memory 1811 without interfering with the operation of the input/output processing.
  • the program code for implementing aspects of the invention described herein may be in the off-chip memory 1803 and be loaded to the on-chip memory 1811 as required.
  • the DSP apparatus shown includes a program memory 1807 including program code 1809 that cause a processor portion 1805 of the DSP apparatus to implement the filtering described herein.
  • An external bus multiplexor 1815 is included for the case that external memory 1803 is required.
  • the term off-chip and on-chip should not be interpreted to imply the there is more than one chip shown.
  • the DSP device 1800 block shown may be provided as a "core" to be included in a chip together with other circuitry.
  • the apparatus shown in FIG. 18 is purely an example.
  • FIG. 19A shows a simplified block diagram of an embodiment of a binauralizing apparatus that is configured to accept five channels of audio information in the form of a left, center and right signals aimed at playback through front speakers, and a left surround and right surround signals aimed at playback via rear speakers.
  • the binauralizer implements binaural filter pairs for each input, including, for the left surround and right surround signals, aspects of the invention so that a listener listening through headphones experiences spatial content while a listener listening to a monophonic mix experiences the signals in a pleasing manner as if from a monophonic source.
  • the binauralizer is implemented using a processing system 1903, e.g., one including a DSP device that includes at least one processor 1905.
  • a memory 1907 is included for holding program code in the form of instructions, and further can hold any needed parameters. When executed, the program code cause the processing system 1903 to execute filtering as described hereinabove.
  • FIG. 19B shows a simplified block diagram of an embodiment of a binauralizing apparatus that accepts four channels of audio information in the form of a left and right from signals aimed at playback through front speakers, and a left rear and right rear signals aimed at playback via rear speakers.
  • the binauralizer implements binaural filter pairs for each input, including for left and right signals, and for the left rear and right rear signals, aspects of the invention so that a listener listening through headphones experiences spatial content while a listener listening to a monophonic mix experiences the signals in a pleasing manner as if from a monophonic source.
  • the binauralizer is implemented using a processing system 1903, e.g., including a DSP device that has a processor 1905.
  • a memory 1907 is included for holding program code 1909 in the form of instructions, and further can hold any needed parameters. When executed, the program code cause the processing system 1903 to execute filtering as described hereinabove.
  • a computer-readable medium is configured with program logic, e.g., a set of instructions that when executed by at least one processor, causes carrying out a set of method steps of methods described herein.
  • program logic e.g., a set of instructions that when executed by at least one processor, causes carrying out a set of method steps of methods described herein.
  • processor may refer to any device or portion of a device that processes electronic data, e.g., from registers and/or memory to transform that electronic data into other electronic data that, e.g., may be stored in registers and/or memory.
  • a "computer” or a “computing machine” or a “computing platform” may include at least one processor.
  • the methodologies described herein are, in one embodiment, performable by one or more processors that accept computer-executable (also called machine-executable) program logic embodied on one or more computer-readable media.
  • the program logic includes a set of instructions that when executed by one or more of the processors carry out at least one of the methods described herein. Any processor capable of executing a set of instructions (sequential or otherwise) that specify actions to be taken are included.
  • processors may include one or more of a CPU, a graphics processing unit, and a programmable DSP unit.
  • the processing system further may include a storage subsystem that includes a memory subsystem including main RAM and/or a static RAM, and/or ROM.
  • the storage subsystem may further include one or more other storage devices.
  • a bus subsystem may be included for communicating between the components.
  • the processing system further may be a distributed processing system with processors coupled by a network. If the processing system requires a display, such a display may be included, e.g., a liquid crystal display (LCD), organic light emitting display, plasma display, a cathode ray tube (CRT) display, and so forth. If manual data entry is required, the processing system also includes an input device such as one or more of an alphanumeric input unit such as a keyboard, a pointing control device such as a mouse, and so forth.
  • LCD liquid crystal display
  • CRT cathode ray tube
  • the processing system in some configurations may include a sound output device, and a network interface device.
  • the storage subsystem thus includes a computer-readable medium that carries program logic (e.g., software) including a set of instructions to cause performing, when executed by one or more processors, one or more of the methods described herein.
  • the program logic may reside in a hard disk, or may also reside, completely or at least partially, within the RAM and/or within the processor during execution thereof by the processing system.
  • the memory and the processor also constitute computer-readable medium on which is encoded program logic, e.g., in the form of instructions.
  • a computer-readable medium may form, or be included in a computer program product.
  • the one or more processors operate as a standalone device or may be connected, e.g., networked to other processor(s), in a networked deployment, the one or more processors may operate in the capacity of a server or a client machine in server-client network environment, or as a peer machine in a peer-to-peer or distributed network environment.
  • the one or more processors may form a personal computer (PC), a tablet PC, a set-top box (STB), a Personal Digital Assistant (PDA), a cellular telephone, a web appliance, a network router, switch or bridge, or any machine capable of executing a set of instructions (sequential or otherwise) that specify actions to be taken by that machine.
  • PC personal computer
  • PDA Personal Digital Assistant
  • each of the methods described herein is in the form of a computer-readable medium configured with a set of instructions, e.g., a computer program that is for execution on one or more processors, e.g., one or more processors that are part of signal processing apparatus.
  • a computer-readable medium configured with a set of instructions, e.g., a computer program that is for execution on one or more processors, e.g., one or more processors that are part of signal processing apparatus.
  • embodiments of the present invention may be embodied as a method, an apparatus such as a special purpose apparatus, an apparatus such as a data processing system, or a computer-readable medium, e.g., a computer program product.
  • the computer-readable medium carries logic including a set of instructions that when executed on one or more processors cause carrying out method steps.
  • aspects of the present invention may take the form of a method, an entirely hardware embodiment, an entirely software embodiment or an embodiment combining software and hardware aspects.
  • the present invention may take the form of program logic, e.g., in a computer readable medium, e.g., a computer program on a computer-readable storage medium, or the computer readable medium configured with computer-readable program code, e.g., a computer program product.
  • While the computer readable medium is shown in an example embodiment to be a single medium, the term “medium” should be taken to include a single medium or multiple media (e.g., a centralized or distributed database, and/or associated caches and servers) that store the one or more sets of instructions.
  • the term "computer readable medium” shall also be taken to include any computer readable medium that is capable of storing, encoding or otherwise configured with a set of instructions for execution by one or more of the processors and that cause the carrying out of any one or more of the methodologies of the present invention.
  • a computer readable medium may take many forms, including but not limited to non-volatile media and volatile media.
  • Non-volatile media includes, for example, optical, magnetic disks, and magneto-optical disks.
  • Volatile media includes dynamic memory, such as main memory.
  • an element described herein of an apparatus embodiment is an example of a means for carrying out the function performed by the element for the purpose of carrying out the invention.
  • any one of the terms comprising, comprised of or which comprises is an open term that means including at least the elements/features that follow, but not excluding others.
  • the term comprising, when used in the claims should not be interpreted as being limitative to the means or elements or steps listed thereafter.
  • the scope of the expression a device comprising A and B should not be limited to devices consisting only of elements A and B.
  • Any one of the terms including or which includes or that includes as used herein is also an open term that also means including at least the elements/features that follow the term, but not excluding others. Thus, including is synonymous with and means comprising.
  • Coupled when used in the claims, should not be interpreted as being limitative to direct connections only.
  • the terms “coupled” and “connected,” along with their derivatives, may be used. It should be understood that these terms are not intended as synonyms for each other.
  • the scope of the expression a device A coupled to a device B should not be limited to devices or systems wherein an output of device A is directly connected to an input of device B. It means that there exists a path between an output of A and an input of B which may be a path including other devices or means.
  • Coupled may mean that two or more elements are either in direct physical or electrical contact, or that two or more elements are not in direct contact with each other but yet still co-operate or interact with each other.

Landscapes

  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Multimedia (AREA)
  • Stereophonic System (AREA)

Claims (12)

  1. Verfahren zum Verarbeiten eines linken Ohrsignals und eines rechten Ohrsignals, um geänderte binaurale Filter zu erzeugen, wobei das Verfahren folgendes umfasst:
    Annehmen (703) eines linken Ohrsignals und eines rechten Ohrsignals, welche die Impulsantworten von entsprechenden binauralen Filtern des linken Ohrs und des rechten Ohrs darstellen, die konfiguriert sind, um ein Audiosignal zu binauralisieren;
    Vermischen (705) des linken Ohrsignals und des rechten Ohrsignals, um ein Summensignal, das zur Summe des linken Ohrsignals und des rechten Ohrsignals proportional ist, und ein Differenzsignal, das zu einer Differenz zwischen dem linken Ohrsignal und dem rechten Ohrsignal proportional ist, zu bilden;
    Filtern (707) des Summensignals mit einem Summenfilter, wobei das Filtern ein gefiltertes Summensignal bildet; Verarbeiten (707) des Differenzsignals mit einem Differenzfilter, wobei die Verarbeitung ein gefiltertes Differenzsignal bildet;
    Entmischen (709) des gefilterten Summensignals und des gefilterten Differenzsignals, um ein geändertes linkes Ohrsignal und ein geändertes rechtes Ohrsignal zu bilden, welche die Impulsantworten von entsprechenden geänderten binauralen Filtern des linken Ohrs und des rechten Ohrs darstellen,
    wobei die geänderten binauralen Filter konfiguriert sind, um ein Audiosignal zu binauralisieren, jeweils durch ein jeweiliges geändertes Summenfilter und ein jeweiliges geändertes Differenzfilter dargestellt werden können und ferner eine linke Ohrausgabe und eine rechte Ohrausgabe aufweisen, wobei jedes Paar von binauralen Filtern jeweils durch ein binaurales Filter eines linken Ohrs und ein binaurales Filter eines rechten Ohrs dargestellt werden kann, wobei jedes Filter eine jeweilige Impulsantwort aufweist, die das Filter charakterisiert, und
    wobei mindestens ein Paar von binauralen Filtern (203-1, 204-1, ..., 203-M, 204-M) konfiguriert ist, um seine jeweilige Audiosignaleingabe zu verräumlichen, um eine direkte Antwort auf einen Zuhörer (107) von einer jeweiligen virtuellen Lautsprecherstelle (209-1, ..., 209-Mv) aus zu integrieren, und um sowohl frühe Echos als auch eine schallharte Antwort eines Hörraums zu integrieren,
    wobei das Verfahren dadurch gekennzeichnet ist:
    dass das Summenfilter zeitlich veränderliche Filtercharakteristiken aufweist,
    dass das Differenzfilter durch das Summenfilter charakterisiert ist, und
    dass für das mindestens eine Paar von binauralen Filtern (203-1, 204-1, ..., 203-M, 204-M) :
    die Zeit-Frequenz-Charakteristiken des Summenfilters (403) anders als die Zeit-Frequenz-Charakteristiken des Differenzfilters (404) sind, wobei die Nachhallzeit des Summenfilters bei allen Frequenzen kleiner ist als jede von: der Nachhallzeit des Differenzfilters, der Nachhallzeit des linken Ohrfilters und der Nachhallzeit des rechten Ohrfilters; und
    die Nachhallzeit des Summenfilters über verschiedene Frequenzen stärker variiert als die jeweilige Variation über Frequenzen der Nachhallzeit des linken Ohrfilters und der Nachhallzeit des rechten Ohrfilters, wobei die Nachhallzeit des Summenfilters mit zunehmender Frequenz abnimmt, wobei für das mindestens eine Paar von binauralen Filtern (203-1, 204-1, ..., 203-M, 204-M) der Übergang der Impulsantwort des Summenfilters auf ihren vernachlässigbaren Pegel im Verlauf der Zeit frequenzabhängig über ein anfängliches Zeitintervall der Impulsantwort des Summenfilters allmählich erfolgt, wobei für das mindestens eine Paar von binauralen Filtern (203-1, 204-1, ..., 203-M, 204-M) der Frequenzinhalt des Summenfilters von anfänglich der vollen Bandbreite auf eine untere Frequenzgrenze über das Übergangszeitintervall abnimmt.
  2. Verfahren nach Anspruch 1, wobei das eine oder die mehreren Audioeingangssignale, das bzw. die durch das Paar von binauralen Filtern (203-1, 204-1, ..., 203-M, 204-M) gefiltert wird/werden, Ausgabesignale erzeugen, die als verräumlicht wahrgenommen werden, wenn sie über Kopfhörer (105) wiedergegeben werden, und als einen geringen wahrgenommenen Nachhall aufweisend wahrgenommen werden, wenn sie nach einer monophonen Mischung, die durch Heruntermischen oder durch Wiedergeben über relativ nah platzierte Lautsprecher erreicht wird, monophon wiedergegeben werden.
  3. Verfahren nach Anspruch 1 oder 2, wobei das geänderte Summensignal geeignet verstärkt wird, um eventuelle Energieverluste in dem geänderten Differenzsignal zu kompensieren, die durch das zeitlich veränderliche Filtern verursacht werden.
  4. Verfahren nach einem der vorhergehenden Ansprüche, wobei für das mindestens eine Paar von binauralen Filtern die binauralen Filtercharakteristiken aus einem Paar von abzugleichenden binauralen Filtercharakteristiken bestimmt werden.
  5. Verfahren nach Anspruch 4, wobei für das mindestens eine Paar von binauralen Filtern die Impulsantwort des Differenzfilters zu späteren Zeitpunkten im Wesentlichen proportional zu dem Differenzfilter des abzugleichenden binauralen Filters ist.
  6. Verfahren nach Anspruch 5, wobei für das mindestens eine Paar von binauralen Filtern die Impulsantwort des Differenzfilters nach 40 ms im Wesentlichen proportional zum Differenzfilter des abzugleichenden binauralen Filters wird.
  7. Verfahren nach einem der Ansprüche 1 bis 6,
    wobei das Ändern des zeitlich veränderlichen Filters durch ein Summenänderungsfilter, das auf ein Signal einwirkt, welches das Summenfilter der abzugleichenden binauralen Filter darstellt, und ein Differenzänderungsfilter, das auf ein Signal einwirkt, welches das Differenzfilter der abzugleichenden binauralen Filter darstellt, dargestellt werden kann,
    wobei das Summenänderungsfilter das Signal im Wesentlichen abschwächt, welches das Summenfilter der abzugleichenden binauralen Filter für Zeitpunkte darstellt, die später als 40 ms sind, und wobei das Differenzänderungsfilter durch die zeitlich veränderlichen Charakteristiken des Summenänderungsfilters definierbar ist.
  8. Verfahren nach Anspruch 7,
    wobei das Summenänderungsfilter durch eine zeitlich veränderliche Impulsantwort zu einem Zeitpunkt, der mit t bezeichnet ist, auf einen Impuls zu dem Zeitpunkt t = τ mit f (t, τ) charakterisierbar ist, und wobei das Summenänderungsfilter ebenfalls durch eine zeitlich veränderliche Frequenzantwort, die eine zeitlich veränderliche Bandbreite umfasst, charakterisierbar ist, wobei die Impulsantwort des Differenzänderungsfilters aus f (t, τ) bestimmbar ist, und wobei die zeitlich veränderliche Bandbreite im Verlauf der Zeit monoton abnimmt.
  9. Verfahren nach Anspruch 8, wobei die zeitlich veränderliche Bandbreite für Zeitpunkte, die größer als ungefähr 40 ms sind, auf weniger als 100 Hz sanft abnimmt.
  10. Verfahren nach Anspruch 8 oder 9,
    wobei die Impulsantwort des Differenzänderungsfilters proportional ist zu √ 2 h D 0 t − √ 2 − 1 ∫ h D 0 t − τ f t τ ⋅ d ⁢ τ .
    Figure imgb0038
  11. Computerlesbarer Datenträger, der eine Programmlogik darin aufweist, die, wenn sie durch mindestens einen Prozessor eines Verarbeitungssystems ausgeführt wird, die Ausführung eines Verfahrens nach einem der vorhergehenden Verfahrensansprüche bewirkt.
  12. Gerät, umfassend:
    ein Verarbeitungssystem, umfassend:
    mindestens einen Prozessor, und
    eine Speichervorrichtung,
    wobei die Speichervorrichtung mit einer Programmlogik konfiguriert ist, die bewirkt, wenn sie ausgeführt wird, dass das Gerät ein Verfahren nach einem der vorhergehenden Verfahrensansprüche ausführt.
EP18155721.6A 2008-09-25 2009-09-15 Binaurale filter für monophonkompatibilität und lautsprecherkompatibilität Active EP3340660B1 (de)

Priority Applications (2)

Application Number Priority Date Filing Date Title
EP23183853.3A EP4274263A3 (de) 2008-09-25 2009-09-15 Binaurale filter für monophone kompatibilität und lautsprecherkompatibilität
EP20159771.3A EP3739908B1 (de) 2008-09-25 2009-09-15 Binaurale filter für monophone kompatibilität und lautsprecherkompatibilität

Applications Claiming Priority (3)

Application Number Priority Date Filing Date Title
US9996708P 2008-09-25 2008-09-25
PCT/US2009/056956 WO2010036536A1 (en) 2008-09-25 2009-09-15 Binaural filters for monophonic compatibility and loudspeaker compatibility
EP09792545.7A EP2329661B1 (de) 2008-09-25 2009-09-15 Binaurale filter für monophonkompatibilität und lautsprecherkompatibilität

Related Parent Applications (2)

Application Number Title Priority Date Filing Date
EP09792545.7A Division EP2329661B1 (de) 2008-09-25 2009-09-15 Binaurale filter für monophonkompatibilität und lautsprecherkompatibilität
EP09792545.7A Division-Into EP2329661B1 (de) 2008-09-25 2009-09-15 Binaurale filter für monophonkompatibilität und lautsprecherkompatibilität

Related Child Applications (2)

Application Number Title Priority Date Filing Date
EP23183853.3A Division EP4274263A3 (de) 2008-09-25 2009-09-15 Binaurale filter für monophone kompatibilität und lautsprecherkompatibilität
EP20159771.3A Division EP3739908B1 (de) 2008-09-25 2009-09-15 Binaurale filter für monophone kompatibilität und lautsprecherkompatibilität

Publications (2)

Publication Number Publication Date
EP3340660A1 EP3340660A1 (de) 2018-06-27
EP3340660B1 true EP3340660B1 (de) 2020-03-04

Family

ID=41346692

Family Applications (4)

Application Number Title Priority Date Filing Date
EP20159771.3A Active EP3739908B1 (de) 2008-09-25 2009-09-15 Binaurale filter für monophone kompatibilität und lautsprecherkompatibilität
EP23183853.3A Pending EP4274263A3 (de) 2008-09-25 2009-09-15 Binaurale filter für monophone kompatibilität und lautsprecherkompatibilität
EP09792545.7A Active EP2329661B1 (de) 2008-09-25 2009-09-15 Binaurale filter für monophonkompatibilität und lautsprecherkompatibilität
EP18155721.6A Active EP3340660B1 (de) 2008-09-25 2009-09-15 Binaurale filter für monophonkompatibilität und lautsprecherkompatibilität

Family Applications Before (3)

Application Number Title Priority Date Filing Date
EP20159771.3A Active EP3739908B1 (de) 2008-09-25 2009-09-15 Binaurale filter für monophone kompatibilität und lautsprecherkompatibilität
EP23183853.3A Pending EP4274263A3 (de) 2008-09-25 2009-09-15 Binaurale filter für monophone kompatibilität und lautsprecherkompatibilität
EP09792545.7A Active EP2329661B1 (de) 2008-09-25 2009-09-15 Binaurale filter für monophonkompatibilität und lautsprecherkompatibilität

Country Status (8)

Country Link
US (1) US8515104B2 (de)
EP (4) EP3739908B1 (de)
JP (1) JP5298199B2 (de)
KR (1) KR101261446B1 (de)
CN (1) CN102165798B (de)
HK (1) HK1256734A1 (de)
TW (1) TWI475896B (de)
WO (1) WO2010036536A1 (de)

Families Citing this family (36)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US9031268B2 (en) 2011-05-09 2015-05-12 Dts, Inc. Room characterization and correction for multi-channel audio
FR2976759B1 (fr) * 2011-06-16 2013-08-09 Jean Luc Haurais Procede de traitement d'un signal audio pour une restitution amelioree.
EP2642407A1 (de) * 2012-03-22 2013-09-25 Harman Becker Automotive Systems GmbH Verfahren zum Abrufen, und System zur Reproduktion eines Audiosignals
WO2013142653A1 (en) * 2012-03-23 2013-09-26 Dolby Laboratories Licensing Corporation Method and system for head-related transfer function generation by linear mixing of head-related transfer functions
JP6160072B2 (ja) * 2012-12-06 2017-07-12 富士通株式会社 オーディオ信号符号化装置および方法、オーディオ信号伝送システムおよび方法、オーディオ信号復号装置
CN104982042B (zh) 2013-04-19 2018-06-08 韩国电子通信研究院 多信道音频信号处理装置及方法
WO2014171791A1 (ko) 2013-04-19 2014-10-23 한국전자통신연구원 다채널 오디오 신호 처리 장치 및 방법
WO2014177202A1 (en) * 2013-04-30 2014-11-06 Huawei Technologies Co., Ltd. Audio signal processing apparatus
DE102013217367A1 (de) * 2013-05-31 2014-12-04 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Vorrichtung und verfahren zur raumselektiven audiowiedergabe
US9319819B2 (en) * 2013-07-25 2016-04-19 Etri Binaural rendering method and apparatus for decoding multi channel audio
KR101782916B1 (ko) * 2013-09-17 2017-09-28 주식회사 윌러스표준기술연구소 오디오 신호 처리 방법 및 장치
US9426300B2 (en) 2013-09-27 2016-08-23 Dolby Laboratories Licensing Corporation Matching reverberation in teleconferencing environments
WO2015048551A2 (en) * 2013-09-27 2015-04-02 Sony Computer Entertainment Inc. Method of improving externalization of virtual surround sound
FR3012247A1 (fr) * 2013-10-18 2015-04-24 Orange Spatialisation sonore avec effet de salle, optimisee en complexite
WO2015060654A1 (ko) 2013-10-22 2015-04-30 한국전자통신연구원 오디오 신호의 필터 생성 방법 및 이를 위한 파라메터화 장치
WO2015099429A1 (ko) 2013-12-23 2015-07-02 주식회사 윌러스표준기술연구소 오디오 신호 처리 방법, 이를 위한 파라메터화 장치 및 오디오 신호 처리 장치
EP3090576B1 (de) 2014-01-03 2017-10-18 Dolby Laboratories Licensing Corporation Verfahren und vorrichtung für die erstellung und die anwendung numerisch optimierter binauraler raumimpulsantworten
CN107750042B (zh) 2014-01-03 2019-12-13 杜比实验室特许公司 响应于多通道音频通过使用至少一个反馈延迟网络产生双耳音频
CN104768121A (zh) 2014-01-03 2015-07-08 杜比实验室特许公司 响应于多通道音频通过使用至少一个反馈延迟网络产生双耳音频
CN108600935B (zh) * 2014-03-19 2020-11-03 韦勒斯标准与技术协会公司 音频信号处理方法和设备
KR101882423B1 (ko) 2014-03-21 2018-08-24 후아웨이 테크놀러지 컴퍼니 리미티드 적어도 제1 쌍의 룸 임펄스 응답에 기초하여, 믹싱 시간 전체를 추정하는 장치 및 방법, 대응하는 컴퓨터 프로그램
KR101856127B1 (ko) 2014-04-02 2018-05-09 주식회사 윌러스표준기술연구소 오디오 신호 처리 방법 및 장치
US10015616B2 (en) 2014-06-06 2018-07-03 University Of Maryland, College Park Sparse decomposition of head related impulse responses with applications to spatial audio rendering
US9560464B2 (en) * 2014-11-25 2017-01-31 The Trustees Of Princeton University System and method for producing head-externalized 3D audio through headphones
EP4447494A2 (de) 2015-02-12 2024-10-16 Dolby Laboratories Licensing Corporation Kopfhörervirtualisierung
CN109565633B (zh) * 2016-04-20 2022-02-11 珍尼雷克公司 有源监听耳机及其双声道方法
CN107358962B (zh) * 2017-06-08 2018-09-04 腾讯科技（深圳）有限公司 音频处理方法及音频处理装置
US10511909B2 (en) 2017-11-29 2019-12-17 Boomcloud 360, Inc. Crosstalk cancellation for opposite-facing transaural loudspeaker systems
FR3075443A1 (fr) * 2017-12-19 2019-06-21 Orange Traitement d'un signal monophonique dans un decodeur audio 3d restituant un contenu binaural
CN108156561B (zh) * 2017-12-26 2020-08-04 广州酷狗计算机科技有限公司 音频信号的处理方法、装置及终端
CN111630877B (zh) 2018-01-29 2022-05-10 索尼公司 声音处理装置、声音处理方法和程序
CN118574070A (zh) 2018-06-12 2024-08-30 奇跃公司 低频信道间相干性控制
WO2020216459A1 (en) * 2019-04-23 2020-10-29 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Apparatus, method or computer program for generating an output downmix representation
US11533560B2 (en) 2019-11-15 2022-12-20 Boomcloud 360 Inc. Dynamic rendering device metadata-informed audio enhancement system
EP3840405A1 (de) * 2019-12-16 2021-06-23 M.U. Movie United GmbH Verfahren und system zur übermittlung und wiedergabe akustischer informationen
CN113613143B (zh) * 2021-07-08 2023-06-13 北京小唱科技有限公司 适用于移动终端的音频处理方法、装置及存储介质

Family Cites Families (52)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US4955057A (en) * 1987-03-04 1990-09-04 Dynavector, Inc. Reverb generator
JPH06121394A (ja) 1992-10-02 1994-04-28 Toshiba Corp 音声出力装置
JPH06165298A (ja) * 1992-11-24 1994-06-10 Nissan Motor Co Ltd 音響再生装置
JP2897586B2 (ja) * 1993-03-05 1999-05-31 ヤマハ株式会社 音場制御装置
US5371799A (en) * 1993-06-01 1994-12-06 Qsound Labs, Inc. Stereo headphone sound source localization system
EP0641139B1 (de) 1993-07-13 2002-09-11 Hewlett-Packard Company, A Delaware Corporation Kombinieren von Ton- und Fernsprech-Daten für einem Rechner
JP3637596B2 (ja) * 1994-01-27 2005-04-13 ソニー株式会社 オーディオ再生装置およびヘッドホン
US5436975A (en) * 1994-02-02 1995-07-25 Qsound Ltd. Apparatus for cross fading out of the head sound locations
US5596644A (en) * 1994-10-27 1997-01-21 Aureal Semiconductor Inc. Method and apparatus for efficient presentation of high-quality three-dimensional audio
US5943427A (en) * 1995-04-21 1999-08-24 Creative Technology Ltd. Method and apparatus for three dimensional audio spatialization
GB9606814D0 (en) * 1996-03-30 1996-06-05 Central Research Lab Ltd Apparatus for processing stereophonic signals
US6009178A (en) * 1996-09-16 1999-12-28 Aureal Semiconductor, Inc. Method and apparatus for crosstalk cancellation
US5809149A (en) * 1996-09-25 1998-09-15 Qsound Labs, Inc. Apparatus for creating 3D audio imaging over headphones using binaural synthesis
US6421446B1 (en) * 1996-09-25 2002-07-16 Qsound Labs, Inc. Apparatus for creating 3D audio imaging over headphones using binaural synthesis including elevation
US5912976A (en) * 1996-11-07 1999-06-15 Srs Labs, Inc. Multi-channel audio enhancement system for use in recording and playback and methods for providing same
JPH1188994A (ja) 1997-09-04 1999-03-30 Matsushita Electric Ind Co Ltd 音像定位装置及び音像制御方法
US6198826B1 (en) * 1997-05-19 2001-03-06 Qsound Labs, Inc. Qsound surround synthesis from stereo
US6067361A (en) * 1997-07-16 2000-05-23 Sony Corporation Method and apparatus for two channels of sound having directional cues
WO1999014983A1 (en) * 1997-09-16 1999-03-25 Lake Dsp Pty. Limited Utilisation of filtering effects in stereo headphone devices to enhance spatialization of source around a listener
CA2325482C (en) 1998-03-25 2009-12-15 Lake Technology Limited Audio signal processing method and apparatus
US6990205B1 (en) * 1998-05-20 2006-01-24 Agere Systems, Inc. Apparatus and method for producing virtual acoustic sound
US6590983B1 (en) * 1998-10-13 2003-07-08 Srs Labs, Inc. Apparatus and method for synthesizing pseudo-stereophonic outputs from a monophonic input
JP4499206B2 (ja) * 1998-10-30 2010-07-07 ソニー株式会社 オーディオ処理装置及びオーディオ再生方法
TW437256B (en) * 1999-03-12 2001-05-28 Ind Tech Res Inst Apparatus and method for virtual sound enhancement
AU2001261344A1 (en) * 2000-05-10 2001-11-20 The Board Of Trustees Of The University Of Illinois Interference suppression techniques
US7583805B2 (en) * 2004-02-12 2009-09-01 Agere Systems Inc. Late reverberation-based synthesis of auditory scenes
US20030035553A1 (en) 2001-08-10 2003-02-20 Frank Baumgarte Backwards-compatible perceptual coding of spatial cues
JP4130779B2 (ja) * 2003-03-13 2008-08-06 パイオニア株式会社 音場制御システム及び音場制御方法
US20040213415A1 (en) * 2003-04-28 2004-10-28 Ratnam Rama Determining reverberation time
US7522733B2 (en) 2003-12-12 2009-04-21 Srs Labs, Inc. Systems and methods of spatial image enhancement of a sound source
US20050147261A1 (en) * 2003-12-30 2005-07-07 Chiang Yeh Head relational transfer function virtualizer
EP1571768A3 (de) * 2004-02-26 2012-07-18 Yamaha Corporation Mischgerät und Tonsignalverarbeitungsverfahren
CN1965610A (zh) 2004-06-08 2007-05-16 皇家飞利浦电子股份有限公司 编码混响声音信号
TWI249361B (en) * 2004-09-21 2006-02-11 Formosa Ind Computing Inc Cross-talk Cancellation System of multiple sound channels
US7634092B2 (en) * 2004-10-14 2009-12-15 Dolby Laboratories Licensing Corporation Head related transfer functions for panned stereo audio content
EP1803115A2 (de) * 2004-10-15 2007-07-04 Koninklijke Philips Electronics N.V. System und verfahren zum verarbeiten von audiodaten zur erzeugung von hall
NO328256B1 (no) 2004-12-29 2010-01-18 Tandberg Telecom As Audiosystem
EP1905004A2 (de) 2005-05-26 2008-04-02 LG Electronics Inc. Verfahren zum codieren und decodieren eines audiosignals
US8331603B2 (en) * 2005-06-03 2012-12-11 Nokia Corporation Headset
JP5108767B2 (ja) 2005-08-30 2012-12-26 エルジー エレクトロニクス インコーポレイティド オーディオ信号をエンコーディング及びデコーディングするための装置とその方法
CN101263739B (zh) * 2005-09-13 2012-06-20 Srs实验室有限公司 用于音频处理的系统和方法
KR100739776B1 (ko) * 2005-09-22 2007-07-13 삼성전자주식회사 입체 음향 생성 방법 및 장치
KR100636252B1 (ko) * 2005-10-25 2006-10-19 삼성전자주식회사 공간 스테레오 사운드 생성 방법 및 장치
KR100708196B1 (ko) * 2005-11-30 2007-04-17 삼성전자주식회사 모노 스피커를 이용한 확장된 사운드 재생 장치 및 방법
CN101401455A (zh) * 2006-03-15 2009-04-01 杜比实验室特许公司 使用子带滤波器的立体声呈现技术
US9100765B2 (en) 2006-05-05 2015-08-04 Creative Technology Ltd Audio enhancement module for portable media player
US8619998B2 (en) * 2006-08-07 2013-12-31 Creative Technology Ltd Spatial audio enhancement processing method and apparatus
TW200743871A (en) * 2006-05-29 2007-12-01 Kenmos Technology Co Ltd Combination of a light source for a direct-type backlight module
US7876903B2 (en) * 2006-07-07 2011-01-25 Harris Corporation Method and apparatus for creating a multi-dimensional communication space for use in a binaural audio system
US8391504B1 (en) * 2006-12-29 2013-03-05 Universal Audio Method and system for artificial reverberation employing dispersive delays
EP1962559A1 (de) * 2007-02-21 2008-08-27 Harman Becker Automotive Systems GmbH Objektive Quantifizierung einer Hörquellenweite eines Lautsprecher-Raumsystems
US8046214B2 (en) * 2007-06-22 2011-10-25 Microsoft Corporation Low complexity decoder for complex transform coding of multi-channel sound

Non-Patent Citations (1)

* Cited by examiner, † Cited by third party
Title
None *

Also Published As

Publication number Publication date
US8515104B2 (en) 2013-08-20
KR101261446B1 (ko) 2013-05-10
TW201031234A (en) 2010-08-16
JP5298199B2 (ja) 2013-09-25
EP2329661B1 (de) 2018-03-21
HK1256734A1 (zh) 2019-10-04
WO2010036536A1 (en) 2010-04-01
KR20110074566A (ko) 2011-06-30
CN102165798B (zh) 2013-07-17
JP2012503943A (ja) 2012-02-09
EP4274263A3 (de) 2024-01-24
EP2329661A1 (de) 2011-06-08
EP3739908B1 (de) 2023-07-12
EP3739908A1 (de) 2020-11-18
US20110170721A1 (en) 2011-07-14
TWI475896B (zh) 2015-03-01
EP3340660A1 (de) 2018-06-27
CN102165798A (zh) 2011-08-24
EP4274263A2 (de) 2023-11-08

Similar Documents

Publication Publication Date Title
EP3340660B1 (de) Binaurale filter für monophonkompatibilität und lautsprecherkompatibilität
US12028701B2 (en) Methods and systems for designing and applying numerically optimized binaural room impulse responses
JP4856653B2 (ja) 被送出チャネルに基づくキューを用いる空間オーディオのパラメトリック・コーディング
JP4944245B2 (ja) 強化された知覚的品質を備えたステレオ信号を生成する方法及び装置
JP5106115B2 (ja) オブジェクト・ベースのサイド情報を用いる空間オーディオのパラメトリック・コーディング
US7583805B2 (en) Late reverberation-based synthesis of auditory scenes
RU2361185C2 (ru) Устройство и способ для формирования многоканального выходного сигнала
CN101044794B (zh) 用于双声道提示码编码方案和类似方案的散射声音整形的方法和设备
US8553895B2 (en) Device and method for generating an encoded stereo signal of an audio piece or audio datastream
JP6377249B2 (ja) オーディオ信号の強化のための装置と方法及び音響強化システム
NO338919B1 (no) Individuell kanalforming for BCC-fremgangsmåter og desslike.
KR20080078882A (ko) 입체 오디오 신호 디코딩

Legal Events

Date Code Title Description
PUAI Public reference made under article 153(3) epc to a published international application that has entered the european phase

Free format text: ORIGINAL CODE: 0009012

STAA Information on the status of an ep patent application or granted ep patent

Free format text: STATUS: THE APPLICATION HAS BEEN PUBLISHED

AC Divisional application: reference to earlier application

Ref document number: 2329661

Country of ref document: EP

Kind code of ref document: P

AK Designated contracting states

Kind code of ref document: A1

Designated state(s): AT BE BG CH CY CZ DE DK EE ES FI FR GB GR HR HU IE IS IT LI LT LU LV MC MK MT NL NO PL PT RO SE SI SK SM TR

STAA Information on the status of an ep patent application or granted ep patent

Free format text: STATUS: REQUEST FOR EXAMINATION WAS MADE

17P Request for examination filed

Effective date: 20181220

RBV Designated contracting states (corrected)

Designated state(s): AT BE BG CH CY CZ DE DK EE ES FI FR GB GR HR HU IE IS IT LI LT LU LV MC MK MT NL NO PL PT RO SE SI SK SM TR

GRAP Despatch of communication of intention to grant a patent

Free format text: ORIGINAL CODE: EPIDOSNIGR1

STAA Information on the status of an ep patent application or granted ep patent

Free format text: STATUS: GRANT OF PATENT IS INTENDED

REG Reference to a national code

Ref country code: HK

Ref legal event code: DE

Ref document number: 1256734

Country of ref document: HK

RIC1 Information provided on ipc code assigned before grant

Ipc: H04S 3/00 20060101ALN20190904BHEP

Ipc: H04S 7/00 20060101AFI20190904BHEP

INTG Intention to grant announced

Effective date: 20190920

GRAS Grant fee paid

Free format text: ORIGINAL CODE: EPIDOSNIGR3

GRAA (expected) grant

Free format text: ORIGINAL CODE: 0009210

STAA Information on the status of an ep patent application or granted ep patent

Free format text: STATUS: THE PATENT HAS BEEN GRANTED

AC Divisional application: reference to earlier application

Ref document number: 2329661

Country of ref document: EP

Kind code of ref document: P

AK Designated contracting states

Kind code of ref document: B1

Designated state(s): AT BE BG CH CY CZ DE DK EE ES FI FR GB GR HR HU IE IS IT LI LT LU LV MC MK MT NL NO PL PT RO SE SI SK SM TR

REG Reference to a national code

Ref country code: GB

Ref legal event code: FG4D

REG Reference to a national code

Ref country code: CH

Ref legal event code: EP

REG Reference to a national code

Ref country code: AT

Ref legal event code: REF

Ref document number: 1241823

Country of ref document: AT

Kind code of ref document: T

Effective date: 20200315

REG Reference to a national code

Ref country code: DE

Ref legal event code: R096

Ref document number: 602009061386

Country of ref document: DE

REG Reference to a national code

Ref country code: IE

Ref legal event code: FG4D

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: NO

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200604

Ref country code: FI

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200304

REG Reference to a national code

Ref country code: NL

Ref legal event code: MP

Effective date: 20200304

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: BG

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200604

Ref country code: SE

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200304

Ref country code: LV

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200304

Ref country code: GR

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200605

Ref country code: HR

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200304

REG Reference to a national code

Ref country code: LT

Ref legal event code: MG4D

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: NL

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200304

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: LT

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200304

Ref country code: EE

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200304

Ref country code: SM

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200304

Ref country code: SK

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200304

Ref country code: IS

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200704

Ref country code: PT

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200729

Ref country code: ES

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200304

Ref country code: CZ

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200304

Ref country code: RO

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200304

REG Reference to a national code

Ref country code: AT

Ref legal event code: MK05

Ref document number: 1241823

Country of ref document: AT

Kind code of ref document: T

Effective date: 20200304

REG Reference to a national code

Ref country code: DE

Ref legal event code: R097

Ref document number: 602009061386

Country of ref document: DE

PLBE No opposition filed within time limit

Free format text: ORIGINAL CODE: 0009261

STAA Information on the status of an ep patent application or granted ep patent

Free format text: STATUS: NO OPPOSITION FILED WITHIN TIME LIMIT

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: IT

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200304

Ref country code: DK

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200304

Ref country code: AT

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200304

26N No opposition filed

Effective date: 20201207

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: SI

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200304

Ref country code: PL

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200304

REG Reference to a national code

Ref country code: CH

Ref legal event code: PL

REG Reference to a national code

Ref country code: BE

Ref legal event code: MM

Effective date: 20200930

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: LU

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20200915

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: IE

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20200915

Ref country code: BE

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20200930

Ref country code: CH

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20200930

Ref country code: LI

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20200930

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: TR

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200304

Ref country code: MT

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200304

Ref country code: CY

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200304

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: MK

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200304

Ref country code: MC

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20200304

P01 Opt-out of the competence of the unified patent court (upc) registered

Effective date: 20230513

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: DE

Payment date: 20240820

Year of fee payment: 16

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: GB

Payment date: 20240822

Year of fee payment: 16

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: FR

Payment date: 20240820

Year of fee payment: 16